It is known how to transmit voice data arising within the scope of a telephone call with the aid of networks based on what is termed the Internet Protocol. Voice-data transmitting of said kind is referred to also as “Voice-over-IP” (VoIP).
VoIP terminals, which are enabled for said type of voice-data transmitting, meanwhile as a rule offer the same added-value services as are made available by terminals belonging to classical telephony.
One of said added-value services is “Computer Telephony Integration” (CTI). It serves to enhance efficiency during voice transmissions. Very simple applications such as computer-supported call-number dialing as well as others ranging as far as all call-center functions can be offered as services by means of CTI services. CTI basically provides a way of supporting telephone services using computer technology. Apart from providing support for features along with their various call-processing functions, that also includes controlling and monitoring private automatic branch exchanges and call-detail recording.
A CTI platform usually includes fault-resistant servers and supports ITU recommendations H.100 and H.110. Functional features customarily include intelligent, network-enabled call-processing controlling as well as the automating of controlling and monitoring functions within a call center, software- and database-controlled functions for automatic call distributing, and mechanisms for logging and displaying stored and evaluated contact data.
Various manufacturer-developed CTI platforms have appeared over the years that have been standardized by different standardizing bodies. There are therefore numerous standards that exhibit a certain mutual dependency. CTI is on the one hand based on known standards such as ISDN and, on the other, defines hardware-structure standards and interface standards. Computer Supported Telecommunications Applications (CSTA) is an instance of an interface standard of said type. CSTA establishes the structure and nature of messages for various service features such as toggling, call diversion, and three-way conferencing.
Methods employed hitherto for providing VoIP services use what is termed the Realtime Transport Protocol (RTP) as that is a protocol for continuously transmitting audio-visual data (streams) over IP-based networks. The protocol was first standardized in 1996 within RFC 1889. A revised RFC was published in 2003. RFC 3550 hence replaces RFC 1889.
RTP is a packet-based protocol and is usually operated over UDP. For IP-telephony applications, the H.323 and SIP protocols are used for negotiating parameters for the call's audio/video streams. H.323 and SIP are therein employed substantially as the signaling protocol between VoIP-enabled terminals (VoIP phones) and to VoIP-enabled computers such as PCs, PDAs, and laptops.
That approach to providing VoIP requires synchronizing between the VoIP protocols and CTI protocols in a VoIP-providing server through additional program-specific measures.